• Journal of Broadcast Engineering
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• v.13 no.1
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• pp.34-42
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• 2008

This paper proposes the mechanism which improves the qualify of video on a limited network bandwidth by applying the ROI technique to an H.264 Scalable Extension technique. The network environment assumed in this parer is the next generation network convergence environment in which the mobile device has one or more network interfaces. Therefore, we allocate the priority to video packets as the hierarchy structure of H.264 SVC-encoded video stream and ROI information, and transmit those packets over appropriate network channel for using those multiple network interfaces. This paper shows two experiments first one is extracting and allocating the video stream on an appropriate network channel, second one is unequal packet transmission by allocated priorities (e.g. ROI). Performance evaluations show that this approach delivers an improved decoded video quality when compared with conventional transmission schemes, especially on device which has multiple network interfaces.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.11 no.6
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• pp.1104-1114
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• 2007

Digital video compression technique has played an important role that enables efficient transmission and storage of multimedia data where bandwidth and storage space are limited. The new video coding standard, H.264/AVC, developed by Joint Video Team(JVT) significantly outperforms previous standards in compression performance. Especially, variable length code(VLC) plays a crucial pun in video and image compression applications. H.264/AVC standard adopted Context-based Adaptive Variable Length Coding(CAVLC) as the entropy coding method. CAVLC of H.264/AVC requires a large number of the memory accesses. This is a serious problem for applications such as DMB and video phone service because of the considerable amount of power that is consumed in accessing the memory. In order to overcome this problem in this paper, we propose a variable length technique that implements memory-free coeff_token, level, and run_before decoding based on arithmetic operations and using only 70% of the required memory at total_zero variable length decoding.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2014.10a
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• pp.527-530
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• 2014

DASH is a well-known streaming technology, which was proposed in 2010 by MPEG and standardized in 2011. Major multimedia contents service providers, including Apple, Microsoft, and Adobe are all using this technology to support their media streaming services. Whenever a new service is requested to the server, the DASH technology helps servicing the multimedia streaming to client by recognizing the capacity of network and by adapting the quality of the multimedia contents. In DASH, the quality of multimedia contents will be automatically lowered to meet the fluctuating network status, when undesirable breaks interrupt the network. In this paper, we classified and analysed the advantages and disadvantages of DASH researches in three aspects: bit-rate measurement method, bandwidth aggregation method; rate adaptation metrics, algorithms and logics; user's experiences and QoE.

• Journal of Broadcast Engineering
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• v.27 no.6
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• pp.914-922
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• 2022

In this paper, we construct a commercial 4K UHD(Ultra High Definition) streaming device that utilizes a 5G mobile communication network as a transport channel and conduct a streaming performance test. It uses RTP(Realtime Transport Protocol) which has transmission quality monitoring capability as a transmission protocol to apply adaptive streaming. In addition, it provides the function to adjust the encoding rate of the video signal so that encoding can be optimized for the change in the bandwidth of the transmission channel. Through the performance test, it is confirmed that the H.265 encoding rate for 4K UHD signal is 48.69Mbps, the average glass-to-glass delay time is 293.60ms, and the average time difference between video and audio for lip sync is 120ms. With the result of performance test, it is shown that the streaming device is applied to 4K UHD Streaming device over 5G mobile communication network.

• Journal of KIISE:Databases
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• v.32 no.4
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• pp.441-451
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• 2005

HTML is mostly stored in native file systems instead of specialized repositories such as a database. Like HTML, XML, the standard for the exchange and the representation of data in the Internet, is mostly resident on native file systems. However. since XML data is irregular and verbose, the disk space and the network bandwidth are wasted compared to those of regularly structured data. To overcome this inefficiency of XML data, the research on the compression of XML data has been conducted. Among recently proposed XML compression techniques, some techniques do not support querying compressed data, while other techniques which support querying compressed data blindly encode data values using predefined encoding methods without considering the types of data values which necessitates partial decompression for processing range queries. As a result, the query performance on compressed XML data is degraded. Thus, this research proposes an XML compression technique which supports direct and efficient evaluations of queries on compressed XML data. This XML compression technique adopts an encoding method, called dictionary encoding, to encode each tag of XML data and applies proper encoding methods for encoding data values according to the inferred types of data values. Also, through the implementation and the performance evaluation of the XML compression technique proposed in this research, it is shown that the implemented XML compressor efficiently compresses real-life XML data lets and achieves significant improvements on query performance for compressed XML data.

• The Journal of the Acoustical Society of Korea
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• v.35 no.3
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• pp.167-174
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• 2016

Underwater communication is difficult to increase the communication capacity because the carrier frequency is lower than that of radio communications on land. This is limited to the bandwidth of the signal under the influence of the characteristics of an ocean medium. As the high transmission speed and large transmission capacity have become necessary in the limited frequency range, the studies on MIMO (Multiple Input Multiple Output) communication have been actively carried out. The performance of the MIMO communication is lower than that of the SIMO (Single Input Multiple Output) communication because cross-talk occurs due to multiusers along with inter symbol interference resulting from the channel characteristics such as delay spread and doppler spread. Although the adaptive equalizer considering multi-channels is used to mitigate the influence of the cross-talk, the algorithm is normally complicated. In this paper, time reversal mirror technique with the characteristic of a self-equalization will be applied to simplify the compensation algorithm and relieve the cross-talk in order to improve the communication performance when the signal transmitted from two channels is received over interference on one channel in the same time. In addition, the performance of the MIMO communication based on the time reversal mirror is verified using data from the SAVEX15(Shallow-water Acoustic Variability Experiment 2015) conducted at the northern area of East China Sea in May 2015.

• Journal of the Korean Society for Nondestructive Testing
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• v.25 no.5
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• pp.391-399
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• 2005

Laser-generated ultrasonic inspection system is a non-contact scanning inspection device with high spatial resolution and wide bandwidth. The amplitude of laser-generated ultrasound is varied according to the energy of pulse laser and the surface conditions of an object where the CW measuring laser beam is pointing. In this paper, we correct the generating errors by measuring the energy of pulse laser beam and correct the measuring errors by extracting the gain information of laser interferometer at each time. h dynamic stabilizer is developed to stably scan on the surface of an object for an laser-generated ultrasonic inspection system. The developed system generates ultrasound after adaptively finding the maximum gain time of an laser interferometer and processes the signal in real time after digitization with high speed. In this paper, we describe hardware configuration and control algorithm to build a stable laser-generated ultrasonic inspection system. Also, we confirmed through experiments that the proposed correction method for the generating errors and measuring errors is effective to improve the performance of a system.

• Journal of KIISE:Information Networking
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• v.28 no.3
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• pp.389-405
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• 2001

In these days, we have increasing demands on the real-time services, especially for the multimedia data transmission in both of wired and wireless environments and thus efficient and stable ways of transmitting realtime data are needs. Although RTP is widely used for internet-based realtime applications, it cannot avoid packet losses, due to the use of UDP stack and its underlying layers. In the case of mobile computing applications, the packet losses are more frequent and consecutive because of the limited bandwidth. In this paper, we first statistically analyze the characteristics of packet losses in the wired and wireless communications, based on Gilbert model, and a new packet recovery scheme for realtime data transmission is presented. To reflect the transmission characteristics of the present network environment, our scheme makes the sender to dynamically adjust the amount of redundant information, using the current packet loss characteristic parameters reported by the receiver. Additionally, we use relatively large and discontinuous offset values, which enables us to recover from both of the random and consecutive packet losses. Due to these characteristics, our scheme is suitable for the mobile computing environment where packet loss rates are relatively high and varies rapidly in a wide range. Since our scheme is based on the analytic model form statistics, it can also be used for other network environments. We have implemented the scheme with Mobile IP and RTP/RTCP protocols to experimentally verify its efficiency.

• Proceedings of the Korean Information Science Society Conference
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• 2003.10c
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• pp.298-300
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• 2003

ea­RED(Efficient Adaptive RED)[1][2]는 다수의 TCP 커넥션이 경쟁하는 병목구간에서 인터넷 라우터 버퍼를 능동적으로 관리하는 다양한 AQM(Active Queue Management) 알고리즘 중의 하나로 RED 라우터 버퍼 관리 알고리즘의 성능을 개선한 라우터 버퍼 관리 알고리즘이다. RED 라우터가 TD 라우터와 같은 네트워크 퍼포먼스를 유지하면서 TCP 커넥션 간 페어니스를 향상시키기 위해서는 link bandwidth. active 커넥션 수. congestion level 등에 대한 네트워크 상태를 고려하여 파라미터에 적절한 값을 설정해야만 한다. 문제는 다이내믹하게 변하는 네트워크 상황에 적합한 파라미터 값을 초기에 설정해주는 것이 매우 어렵다는 점이다. [3]. ea­RED는 max threshold와 min threshold 값을 네트워크 상황에 따라 동적으로 조절함으로써 이런 문제를 해결했고, 기존 RED에 비해 라우터 버퍼는 50% 정도만 사용하면서도, 페어니스 인덱스(Fairness Index)[4]가 최대 41.42% 개선되었다. [1] [2] 그러나 송신 TCP 커넥션의 수가 늘어날수록 성능향상에 대한 효과가 감소되었고, 드롭 패킷수가 TD나 RED 라우터 버퍼관리 알고리즘에 비해 많았기 때문에 라우터의 출력(output) 총 패킷 용량이 최대 약 2.3% 정도 TD나 RED 라우터 버퍼관리 알고리즘에 비해 적었다. 이 부분을 개선하기 위해 기존 ea­RED 알고리즘에 LR_Lines 예측 알고리즘을 적용한 ea­RED++ 알고리즘을 구현하였고, 실험 결과 페어니스 인덱스는 기존 ea­RED에 비해 최대 약 30% 정도 향상되었고, 총 output 패킷 용량의 손실률은 최대 50%정도 감소하여 기존 ea­RED에 비해 향상된 성능을 보여주었다.웍스 네트워크상의 다양한 디바이스들간의 네트워크 다양화와 분산화 기능을 얻을 수 있었고, 기존의 고가의 해외 솔루션인 Echelon사의 LonMaker 소프트웨어를 사용하지 않고도 국내의 순수 솔루션인 리눅스 기반의 LonWare 3.0 다중 바인딩 기능을 통해 저 비용으로 홈 네트워크 구성 관리 서버 시스템 개발에 대한 비용을 줄일 수 있다. 기대된다.e 함량이 대체로 높게 나타났다. 점미가 수가용성분에서 goucose대비 용출함량이 고르게 나타나는 경향을 보였고 흑미는 알칼리가용분에서 glucose가 상당량(0.68%) 포함되고 있음을 보여주었고 arabinose(0.68%), xylose(0.05%)도 다른 종류에 비해서 다량 함유한 것으로 나타났다. 흑미는 총식이섬유 함량이 높고 pectic substances, hemicellulose, uronic acid 함량이 높아서 콜레스테롤 저하 등의 효과가 기대되며 고섬유식품으로서 조리 특성 연구가 필요한 것으로 사료된다.리하였다. 얻어진 소견(所見)은 다음과 같았다. 1. 모년령(母年齡), 임신회수(姙娠回數), 임신기간(姙娠其間), 출산시체중등(出産時體重等)의 제요인(諸要因)은 주산기사망(周産基死亡)에 대(對)하여 통계적(統計的)으로 유의(有意)한 영향을 미치고 있어 $25{\sim}29$세(歲)의 연령군에서, 2번째 임신과 2번째의 출산에서 그리고 만삭의 임신 기간에, 출산시체중(出産時體重) $3.50{\sim}3.99kg$사이의 아이에서 그 주산기사망률(周産基死亡率)이 각각 가장 낮았다. 2. 사산(死産)과 초생아사망(初生兒死亡)을 구분(區分)하여 고려해 볼때 사산(死産)은 모성(母性)의 임신력(姙娠歷)과 매우 밀접한 관련이 있는 것으로 사료(思料)되었고 초생아사망(初生兒死亡)은 미숙아(未熟兒)와 이에 관련된 병

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.43 no.10 s.352
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• pp.79-89
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• 2006

TCP is the most widely used transport protocol in Internet applications that guarantees a reliable data transfer. But, in the wireless multi-hop networks, TCP performance is degraded because it is designed for wired networks. The main reasons of TCP performance degradation are contention for wireless medium at the MAC layer, hidden terminal problem, exposed terminal problem, packet losses in the link layer, unfairness problem, reordering problem caused by path disconnection, bandwidth waste caused by exponential backoff of retransmission timer due to node's mobility and so on. Specially, in the mobile ad-hoc networks, discrepancy between a station's transmission range and interference range produces hidden terminal problem that decreases TCP performance greatly by limiting simultaneous transmission at a time. In this paper, we propose a new MAC algorithm for mobile ad-hoc networks to solve the problem that a node can not transmit and just increase CW by hidden terminal. In the IEEE 802.11 MAC DCF, a node increases CW exponentially when it fails to transmit, but the proposed algorithm, changes CW adaptively according to the reason of failure so we get a TCP performance enhancement. We show by ns-2 simulation that the proposed algorithm enhances the TCP performance by fairly distributing the transmission opportunity to the failed nodes by hidden terminal problems.